support send_timeout

examples/webrtc.rs

@@ -8,10 +8,10 @@ mod webrtc_dummy;
 use crate::webrtc_dummy::WebRTCStream;
 
 use std::io::Write;
-use bytes::Bytes;
 
-use clap::{Arg, Command};
 use anyhow::Result;
+use bytes::Bytes;
+use clap::{Arg, Command};
 use tokio::time::Duration;
 
 use webrtc::peer_connection::math_rand_alpha;
@@ -75,7 +75,10 @@ async fn main() -> Result<()> {
         // Wait for the answer to be pasted
         println!(
             "Start new terminal run: \n{} \ncopy remote endpoint and paste here",
-            format!("cargo r --features webrtc --example webrtc -- --offer {}", local_endpoint)
+            format!(
+                "cargo r --features webrtc --example webrtc -- --offer {}",
+                local_endpoint
+            )
         );
         // readline blocking
         let line = std::io::stdin()
@@ -84,7 +87,10 @@ async fn main() -> Result<()> {
             .ok_or_else(|| anyhow::anyhow!("No input received"))??;
         webrtc_stream.set_remote_endpoint(&line).await?;
     } else {
-        println!("Copy local endpoint and paste to the other peer: \n{}", local_endpoint);
+        println!(
+            "Copy local endpoint and paste to the other peer: \n{}",
+            local_endpoint
+        );
     }
 
     let s1 = webrtc_stream.clone();

src/webrtc.rs

@@ -1,31 +1,29 @@
-use std::sync::Arc;
-use std::net::{IpAddr, Ipv4Addr, SocketAddr};
-use std::io::{Error, ErrorKind};
-use std::time::Duration;
 use std::collections::HashMap;
+use std::io::{Error, ErrorKind};
+use std::net::{IpAddr, Ipv4Addr, SocketAddr};
+use std::sync::Arc;
+use std::time::Duration;
 
-use webrtc::api::APIBuilder;
 use webrtc::api::setting_engine::SettingEngine;
+use webrtc::api::APIBuilder;
 use webrtc::data_channel::RTCDataChannel;
+use webrtc::ice::mdns::MulticastDnsMode;
 use webrtc::ice_transport::ice_server::RTCIceServer;
-use webrtc::peer_connection::RTCPeerConnection;
 use webrtc::peer_connection::configuration::RTCConfiguration;
 use webrtc::peer_connection::peer_connection_state::RTCPeerConnectionState;
 use webrtc::peer_connection::sdp::session_description::RTCSessionDescription;
-use webrtc::ice::mdns::MulticastDnsMode;
+use webrtc::peer_connection::RTCPeerConnection;
 
-use crate::{
-    protobuf::Message,
-    sodiumoxide::crypto::secretbox::Key,
-    ResultType,
-};
-
-use base64::Engine;
 use base64::engine::general_purpose::STANDARD as BASE64_STANDARD;
+use base64::Engine;
 use bytes::{Bytes, BytesMut};
-use tokio::time::timeout;
 use tokio::sync::watch;
 use tokio::sync::Mutex;
+use tokio::time::timeout;
+
+use crate::protobuf::Message;
+use crate::sodiumoxide::crypto::secretbox::Key;
+use crate::ResultType;
 
 pub struct WebRTCStream {
     pc: Arc<RTCPeerConnection>,
@@ -53,29 +51,36 @@ impl Clone for WebRTCStream {
 }
 
 impl WebRTCStream {
-
+    #[inline]
-    pub fn get_remote_offer(endpoint: &str) -> ResultType<String> {
+    fn get_remote_offer(endpoint: &str) -> ResultType<String> {
         // Ensure the endpoint starts with the "webrtc://" prefix
         if !endpoint.starts_with("webrtc://") {
-            return Err(Error::new(ErrorKind::InvalidInput, "Invalid WebRTC endpoint format").into());
+            return Err(
+                Error::new(ErrorKind::InvalidInput, "Invalid WebRTC endpoint format").into(),
+            );
         }
 
         // Extract the Base64-encoded SDP part
         let encoded_sdp = &endpoint["webrtc://".len()..];
         // Decode the Base64 string
-        let decoded_bytes = BASE64_STANDARD.decode(encoded_sdp).map_err(|_|
+        let decoded_bytes = BASE64_STANDARD
-            Error::new(ErrorKind::InvalidInput, "Failed to decode Base64 SDP")
+            .decode(encoded_sdp)
-        )?;
+            .map_err(|_| Error::new(ErrorKind::InvalidInput, "Failed to decode Base64 SDP"))?;
         Ok(String::from_utf8(decoded_bytes).map_err(|_| {
-            Error::new(ErrorKind::InvalidInput, "Failed to convert decoded bytes to UTF-8")
+            Error::new(
+                ErrorKind::InvalidInput,
+                "Failed to convert decoded bytes to UTF-8",
+            )
         })?)
     }
 
-    pub fn sdp_to_endpoint(sdp: &str) -> String {
+    #[inline]
+    fn sdp_to_endpoint(sdp: &str) -> String {
         let encoded_sdp = BASE64_STANDARD.encode(sdp);
         format!("webrtc://{}", encoded_sdp)
     }
 
+    #[inline]
     async fn get_key_for_peer(pc: &Arc<RTCPeerConnection>) -> String {
         if let Some(local_desc) = pc.local_description().await {
             if local_desc.sdp_type != webrtc::peer_connection::sdp::sdp_type::RTCSdpType::Offer {
@@ -89,10 +94,7 @@ impl WebRTCStream {
         "".into()
     }
 
-    pub async fn new(
+    pub async fn new(remote_endpoint: &str, ms_timeout: u64) -> ResultType<Self> {
-        remote_endpoint: &str,
-        ms_timeout: u64,
-    ) -> ResultType<Self> {
         log::debug!("New webrtc stream to endpoint: {}", remote_endpoint);
         let remote_offer = if remote_endpoint.is_empty() {
             "".into()
@@ -115,9 +117,7 @@ impl WebRTCStream {
         s.set_ice_multicast_dns_mode(MulticastDnsMode::Disabled);
 
         // Create the API object
-        let api = APIBuilder::new()
+        let api = APIBuilder::new().with_setting_engine(s).build();
-            .with_setting_engine(s)
-            .build();
 
         // Prepare the configuration
         let config = RTCConfiguration {
@@ -158,16 +158,25 @@ impl WebRTCStream {
             let pc_for_close2 = pc_for_close.clone();
             Box::pin(async move {
                 log::debug!("Peer connection state : {}", s);
-                if s == RTCPeerConnectionState::Disconnected {
+                match s {
+                    RTCPeerConnectionState::Disconnected
+                    | RTCPeerConnectionState::Failed
+                    | RTCPeerConnectionState::Closed => {
                         let _ = on_connection_notify2.send(true);
                         log::debug!("WebRTC session closing due to disconnected");
                         let _ = stream_for_close2.lock().await.close().await;
                         log::debug!("WebRTC session stream closed");
-                } else if s == RTCPeerConnectionState::Failed || s == RTCPeerConnectionState::Closed {
+
                         let mut lock = SESSIONS.lock().await;
                         let key = WebRTCStream::get_key_for_peer(&pc_for_close2).await;
-                    log::debug!("WebRTC session removing key from cache: {}", key);
                         lock.remove(&key);
+                        log::debug!(
+                            "WebRTC session removed key from cache: {} current len: {}",
+                            key,
+                            lock.len()
+                        );
+                    }
+                    _ => {}
                 }
             })
         }));
@@ -287,7 +296,26 @@ impl WebRTCStream {
     }
 
     pub async fn send_bytes(&mut self, bytes: Bytes) -> ResultType<()> {
+        if self.send_timeout > 0 {
+            match timeout(
+                Duration::from_millis(self.send_timeout),
+                self.wait_for_connect_result(),
+            )
+            .await
+            {
+                Ok(_) => {}
+                Err(_) => {
+                    self.pc.close().await.ok();
+                    return Err(Error::new(
+                        ErrorKind::TimedOut,
+                        "WebRTC send wait for connect timeout",
+                    )
+                    .into());
+                }
+            }
+        } else {
             self.wait_for_connect_result().await;
+        }
         let stream = self.stream.lock().await.clone();
         stream.send(&bytes).await?;
         Ok(())
@@ -339,6 +367,5 @@ pub fn is_webrtc_endpoint(endpoint: &str) -> bool {
 #[cfg(test)]
 mod tests {
     #[test]
-    fn test_dc() {
+    fn test_dc() {}
-    }
 }